The invention concerns a method for encoding and/or decoding voice signals, in particular for digital dictating devices.
For encoding, a voice signal is usually initially low-pass filtered at a limiting frequency of less than 4 kHz and the resulting signal is sampled at a sampling rate of 8 kHz. The sampled signal is converted into a digital voice signal consisting of a sequence of voice signal sampling values. Prediction parameters are determined from this sequence of voice signal sampling values for use in the voice signal encoder and decoder. Moreover, for each voice signal sampling value a predicted value is calculated using the above-mentioned prediction parameters for each of the voice signal sampling values. The difference between each signal sampling value and its predicted value is quantized, digitally encoded and passed, modulated together with the prediction parameters, to a storage medium which may be e.g. a magnetic tape or a RAM memory. The signal stored by the storage medium is divided into its individual partial signals and used in a voice decoder to reproduce the original voice signal as precisely as possible.
Conventional methods operating according to the above-mentioned basic principles are disclosed in the patent documents U.S. Pat. No. 4, 133,976, U.S. Pat. No. 3,631 ,520 and
EP-A0657874 describes a voice signal encoder which calculates prediction parameters from a digitized voice signal. An adaptive code book is used to determine an excitation signal component. In addition, multiple pulse components of the excitation signal are determined using the voice signal. For processing, the voice signals are divided into varions time regions and subjected to individual further processing.
U.S. Pat. No. 5,327,520 discloses a voice encoder with which, using a backward adaptive AGC, stored code vectors are evaluated for comparison to input voice signals. For simplification, they are administered in tables.
The publication "Low Complexity Speed Coder for Personal Multimedia Communication", J. Ikedo et al., 1995 fourth IEEE International Conference on Universal Personal Communications Record, Gateway to the 21.sup.st Century, Tokyo 06 to Nov. 10, 1995, describes an adaptive code book having entries from a delayed overall excitation signal. In this code book, each inital subblock is completely examined. Only a defined partial region is searched in each of the additional subblocks.
The publication "Efficient Computation and Encoding of the Multipulse Excitation for LPC", M. Berouti et al., ICASSP 84.sup.th Procedings of the IEEE International Conference on Acoustics, Speech, and signal Processing, San Diego, USA, Mar. 23-25, 1984, pages 10.1/1-4, describes a coding procedure with which multipulse excitation vectors are encoded using their pulse positions and associated amplitudes.
Departing from this prior art, the underlying purpose of the invention is to improve the quality of reproduction of a voice signal recorded with a digital dictating device.